The System portion of the Altus Control Center contains a variety of global status and configuration pages, each described below. The System menu is found via the Altus Control Center which is accessible by launching your preferred web browser on a computer connected to your studio network and entering the IP address of the Altus in the address bar.
Status
Clicking on the Status menu item in the System section of the Altus Control Center displays the current Version as well as various system information.
Backup / Restore
Configuration of profiles can be backed up in an XML formatted file. Pressing the Backup settings button will download the current system to the computer.
To restore a backup, select a file with the choose file button followed by pressing the Restore settings button.
Licensing
Altus comes equipped with base level of fader strips. Additional groups of four faders – up to 24 in total – can be added by way of a licensing
Click on Licensing. The License Management page shows the current license key numbers (if any) and the total number of activated features.
Paste a new license key in the text box, then click the Add License button to enable the additional faders.
The Release button will release the use of all license keys so that they can be transferred to a different instance. If a License is in an inactive state, the Activate button can be used to register the use of any licenses listed.
Each License key will have a Remove button which removes the license from the product. Best to Release prior to removing to make sure the license was released so it can be used again.
Passwords
The Configuration pages password allows you to set a password associated with the “user” log-in. This access level allows not only access to the console surface itself but to the Altus control center. This is typically reserved for the engineering staff. Type the password in the “Password” field, re-type it in the “Confirmation” field, then press the “Change” button to save.
The Surface access password allows you to set a password associated with the “surface” log-in. This access level only allows access to the console surface itself. This is typically the log-in for operators. Type the password in the “Password” field, re-type it in the “Confirmation” field, then press the “Change” button to save.
Customize
Various special configurations can be here based on if a license enables the functionality. For example, if a webRTC function is enabled (through a license), there will be a field to populate the webRTC signaling server URL. This is a server that helps establish connection for both ends of a webRTC connection.
Network interface option is present for systems that have multiple network interfaces and to define a bonding for Audio packets instead of allowing the boot up process to select on random.
Internal DSP task period should be left at 1ms for best performance.
System Log
A log of the last 100 system events is available by clicking on Log in the Diagnostic menu.
Log History
A history of the system log is available by clicking Log History.
Clicking an individual log file opens the file in the browser window. Clicking on the download icon downloads the log file to your computer’s default download folder. By default, Windows PCs use Notepad to open .log files while Macs use Console.
To delete log files, select the files to be deleted then click the Delete Selected Files button. The system maintains logs so that a single file max size is 1Mb. No more than 10 files will be stored.
Log information can also be pushed to a external syslog server if desired. Enter in an IP value of the server. The Severity level allows for adjustment to the amount of logging to occur. Not recommended to use Debug level in a production environment as the excessive amount of information could impact system performance.
Stream Status
Stream status page shows the condition of streams being received into the product.
BufferMin - (in audio samples) The minimum receive audio buffer remaining, computed at the moment a new packet comes in. Zero is technically ok, but with no margin for any additional network delay. A negative figure is an underflow and means an audio glitch. A typical ‘good’ reading will show a moderate minimum buffer occupancy (48 to 100 samples), but neither too low nor too high. Values much larger than zero mean excess buffer is being used, causing excess latency.
RTP ts position min, stddev - This is the relative position of the incoming RTP timestamps compared with local RTP time, in samples. Negative means the packet is carrying audio data from the past. A 'normal' RTP ts position in a perfectly operating system, would be a negative number equal to the number of samples in the packet.
A positive number is, generally speaking, impossible (you can't send a packet and have it arrive in the future), but in practice means there is synchronization error, either in the sending device (the sending device local time clock running fast), or the rx device (rx device local time clock running slow), or a combination.
The most negative RTP timestamp position corresponds to greatest risk of audio buffer underflow (i.e. packets coming in late.) So the minimum is displayed.
The standard deviation of the RTP ts is a measure of the time jitter in the stream. The computation of sigma uses running averages.
When you have both sending and receiving devices using AES67 and PTP clock, there are global time stamps that can be used directly, along with the link offset value. This is called synchronous mode. Synchronous means same time. Syntonous (meaning same tone, or frequency), is what legacy Livewire sync used. Frequency lock, but no common absolute time reference. For syntonous receive mode, used when you cannot rely on the RTP timestamps (for instance legacy Livewire where there are no valid timestamps), the stream receiver automatically picks a stream reference point and starts buffering. The size of the buffer needed then is determined by the jitter in the stream, so that the buffers don't keep underflowing. In practice, the automatic syntonous receive tends to hunt for and find the latest packets, and start the buffering from there.
Stream reset - This counts the number of audio glitches, which usually are caused by buffer underflows. This happens if the link offset is not set high enough, so late packets arrive too late to be used. This counter is cleared when you load a new source into the stream receive channels.